Real-time two-way communication (which may be not just audio only, but also audio and video) can be carried out between two electronic communication devices that are generically referred to here as telephony devices. Such devices have evolved over the years from simple plain old telephone system (POTS) analog wire line stations to cellular network phones, smart mobile phones, voice over IP (VOIP) stations, and desktop and laptop personal computers running VOIP applications. There is a desire to remain backwards compatible with the original, relatively small bandwidth allocated to a voice channel in a POTS network. This in part has prevented the emergence of a “high fidelity” telephone call, despite the availability of such technology.
Modern telephony devices such as smart phones support not only voice communications over a voice channel, but also multimedia services, such as real time audio, video chat, and mobile TV, over a data channel. Improving the sound quality of a downlink audio signal is particularly desirable for smart phones as they may be more susceptible to electromagnetic interference, due to their reliance on cellular wireless links. In addition, smart phones are often used in noisy sound environments, such as outside in the wind or near a busy highway or a crowded people venue.
Smart phones have several stages of audio signal processing that are applied to the downlink audio signal, which is received from the communications network (before the signal is audiblized to a near-end user of the device through a speaker). In addition, signal processing algorithms have been developed to improve the intelligibility of the far-end user's speech contained in the downlink audio signal, when the near-end user is in areas of high ambient noise. Typically, the near-end user will manually adjust the volume, press the device closer to her ear, or wear a headset to overcome ambient noise while receiving the downlink audio signal. An intelligibility boost algorithm will help by automatically adjusting an equalization filter in order to increase the gain at high frequency components relative to the low frequency components of the downlink speech as a function of either a measured ambient noise level or the current user-selected volume setting. This will make the speech more intelligible (albeit slightly artificial sounding).